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AAC beaten at low bitrates, why?
Mac
post Aug 5 2002, 20:01
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Looking at the recent tests on 64kb files, I was wondering how come AAC underperforms compared to OGG, WMA and whatever else? I thought AAC was able to handle low bitrates very well due to it's type of encoding... if so, how come these other formats are beating it? Is it something that can be worked on in future releases from Psytel? Spose they would of been working mainly on high bitrate quality to whoop MPC's ass? :o)
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rjamorim
post Aug 5 2002, 20:06
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QUOTE
Originally posted by Mac
I thought AAC was able to handle low bitrates very well due to it's type of encoding...    if so, how come these other formats are beating it? 


There isn't any high quality AAC implementation available that uses Intendity Stereo (although IS is part of the specs). That's the reason.

AAC+ (AAC+SBR) should be available soon and will address this issue as well.

QUOTE
Is it something that can be worked on in future releases from Psytel?  Spose they would of been working mainly on high bitrate quality to whoop MPC's ass? ohmy.gif)


High bitrate work on Psytel AACenc is nearly done, I would believe.
Ivan is probably working on low bitrate tuning more.

Regards;

Roberto.


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Mac
post Aug 5 2002, 20:57
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Aah cool ohmy.gif) I heard something about AAC+SBR on the Psytel website, but didn't quite understand it!

I take it there's no reason to wait for this new version before encoding my wav files at -extreme? I won't gain any extra quality by waiting?
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Frank Klemm
post Aug 5 2002, 20:59
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QUOTE
Originally posted by rjamorim


There isn't any high quality AAC implementation available that uses Intendity Stereo (although IS is part of the specs). That's the reason.

AAC+ (AAC+SBR) should be available soon and will address this issue as well.

High bitrate work on Psytel AACenc is nearly done, I would believe.
Ivan is probably working on low bitrate tuning more.

Regards;

Roberto.


Castanets encoding with different encoder at 128 kbps. My ratings:

1. Envivo AAC 4.9 (nearly perfect)
2. FhG AAC 5.0-0.2 (most the time better than Envivo, but strange loud clicks in the right channel von time to time)
3. HHI AAC 4.6 (some strange noise)
3. Psytel AAC 4.6 (some strange noise)
5. NEC 4.3 (one castanets was completely wrong, strange noise)
6. Lame 3.91 3.8
7. MPC 1.7.9 3.5 (muffled, strange noise)
8. ISO AAC 3.2 (some strange noise, castanets messed up)
9. tooLame 2.0 (worse noises, problems with the castagnets)
10. Philips AAC 1.0 (weird sound at all)


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Mac
post Aug 5 2002, 21:12
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Hmmm, where do you get all these AAC encoders from? Or is it a case of you don't get them. If not, Ivan needs to make his the best just to teach them a lesson! ohmy.gif)
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rjamorim
post Aug 5 2002, 22:10
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QUOTE
Originally posted by Mac
Hmmm, where do you get all these AAC encoders from?


Just what I was wondering! smile.gif

And I wonder how can Philips sound worse than ISO! ohmy.gif

QUOTE
If not, Ivan needs to make his the best just to teach them a lesson! ohmy.gif)


Well, keep in mind that, while castanets is a good sample to tune encoders, it isn't proper to compare encoders' quality using it. It's an exception sample, because it's not representative of generic music.

For your former question:
You can encode with -extreme safely now. Heh, I have files encoded with AACenc 1.6 -extreme that still sound perfect.

Regards;

Roberto.

PS: Please avoid using : o ) as smiley. wink.gif
Or check "Disable smilies in This Post" before posting.


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Ardax
post Aug 5 2002, 23:41
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QUOTE
Originally posted by rjamorim
You can encode with -extreme safely now. Heh, I have files encoded with AACenc 1.6 -extreme that still sound perfect.


Really? How do you keep your files from sounding worse when new encoder versions come out? I don't know what I'm going to do when Vorbis 1.01 comes out. tongue.gif

Sorry, I just couldn't pass it up. smile.gif
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rjamorim
post Aug 5 2002, 23:57
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QUOTE
Originally posted by Ardax
Really?  How do you keep your files from sounding worse when new encoder versions come out?  I don't know what I'm going to do when Vorbis 1.01 comes out.  tongue.gif


Well, they already sounded very good encoded with 1.6. If they sound OK to my ears, I see no reason to reencode. smile.gif


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layer3maniac
post Aug 6 2002, 04:44
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QUOTE
Originally posted by Mac
Hmmm, where do you get all these AAC encoders from?
Or better yet, where can "I" get these encoders from?
biggrin.gif
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Dibrom
post Aug 6 2002, 05:30
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QUOTE
Originally posted by rjamorim

Well, keep in mind that, while castanets is a good sample to tune encoders, it isn't proper to compare encoders' quality using it. It's an exception sample, because it's not representative of generic music.


One also has to consider what "generic" music truly is. Is my IDM generic music? Is my metal? What about my classical guitar music (filled with transients) like Bernd Steidl?

How do you define generic music? Is it what's on MTV? Is it the latest, greatest pop band? Classic rock?

For the most part, I can accept people saying things are not representative when talking about some admittedly obscure electronic music which a very large majority of the population wouldn't even consider music in the first place, but something like castanets surely doesn't fall in that category. A lot of music has rapid percussive sounds in it, and so a lot of music has the potential to be just "exceptional" as castanets.

Bottom line is how do you decide which samples can be used to compare codecs? It's kind of paradoxical to use samples which all encoders will sound equal on because they are extremely easy to encode. By the same token, any randomly chosen sample which ends up causing particular problems for an encoder (or many) can be said to be non-representative by just about anyone... so how do you decide where to draw the line?

Bit OT, yes, but I just wanted to add a bit of insight here wink.gif
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rjamorim
post Aug 6 2002, 05:54
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QUOTE
Originally posted by Dibrom
A lot of music has rapid percussive sounds in it


Yeah, but not as in castanets. Very few other music has such rapid percussion - and Castanets in nearly only percussion, too.

BESIDES, it's a well know fact that subband coders (MPC) handle these kinds of samples much better than transfom coders (Vorbis, MP3, AAC). So, based on these results, would you assume that MP3, AAC and Vorbis suck?

SO, my point is that, if you want to compare codecs quality, you can't use ONLY castanets as Mr. Klemm used (Although using it to compare encoders of a same format is relatively fair. Relatively, because one encoder can perform very well on every aspect of audio encoding, but wasn't properly tuned to handle transients). It's VERY well known that castanets is unfavorable to transform codecs.

If you want to test codecs quality, it's OK (i think) to include castanets. But you should be sure to include classical, pop, soft, metal, unplugged... as well. Both easy and hard samples. THAT would be a generic representative, IMO.

And, again, not ONLY castanets.

Hope I made myself clear.

Regards;

Roberto.


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Dibrom
post Aug 6 2002, 06:05
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QUOTE
Originally posted by rjamorim


Yeah, but not as in castanets. Very few other music has such rapid percussion - and Castanets in nearly only percussion, too.


Hrmm.. this is what I'm not so sure about. I listen to a lot of music with similar sounds in it. How many other people do as well? I'm just saying... how can you quantify this exactly?

QUOTE
[b]BESIDES, it's a well know fact that subband coders (MPC) handle these kinds of samples much better than transfom coders (Vorbis, MP3, AAC). So, based on these results, would you assume that MP3, AAC and Vorbis suck?


No, not at all. But I would believe it would be valid to make a statement about quality at least in a given situation if results hold true over a variety of similar samples, even if these samples may not be considered by any given person to be representative of generic music.

QUOTE
[b]SO, my point is that, if you want to compare codecs quality, you can't use ONLY castanets as Mr. Klemm used (Although using it to compare encoders of a same format is relatively fair. Relatively, because one encoder can perform very well on every aspect of audio encoding, but wasn't properly tuned to handle transients). It's VERY well known that castanets is unfavorable to transform codecs.


I very much agree with this. 1 sample is surely nowhere enough. What I'm getting at though is what happens when someone wants to makes a comparison based on a slew of difficult samples. Say 20 or 30 of the most well known difficult to encode samples. Can a general statement about quality be made based on these? I think it can. That's at least how I initially did it with --alt-preset vs --r3mix. That's how I tuned things and how I came to the conclusion that the --alt-presets are better. If you can let someone say that 1 sample is not representative though (or worse, not a proper comparison), what's to stop them from saying all 30 are not representative? See what I mean?

QUOTE
[b]If you want to test codecs quality, it's OK (i think) to include castanets. But you should be sure to include classical, pop, soft, metal, unplugged... as well. Both easy and hard samples. THAT would be a generic representative, IMO.

And, again, not ONLY castanets.

Hope I made myself clear.


Certainly smile.gif I'm just wary of making a statement such as "it's not valid", because I think it's a very hard line to draw, and if you begin to make such statements in one situation (you or anyone else, nobody in particular), then it can lead to forming an ideology that critical samples (where the largest differences exist between codecs, IMO the more important situations) are no good for comparison at all.
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rjamorim
post Aug 6 2002, 06:16
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QUOTE
Originally posted by Dibrom
Say 20 or 30 of the most well known difficult to encode samples.  Can a general statement about quality be made based on these?


I think it probably would be valid.

But, instead choosing worst-case samples, I would rather choose samples of different music styles. (If these worst-case samples are "representative" - perfect!)

My rationalization:
If you use samples from different music stiles to tune your encoder, you'll be tuning it for the general public that will be using your encoder.

If you use only problem case samples, you'll be tuning your encoder for listening tests that use only these samples. You'll be tuning it for a limited part of the public that enjoys that kind of music too, but a much wider public would be reached using "generic samples"

After all, what's worth if an encoder performs superbly on Castanets and Fatboy, but fails on Pink Floyd? Such an encoder would be worthless! biggrin.gif

Regards;

Roberto.

BTW: Just to clarify:
When I say "encoder", it's the implementation: Lame, MP3enc, AACenc...
When I say "codec", it's the format: MP3, AAC...


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Dibrom
post Aug 6 2002, 06:20
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QUOTE
Originally posted by rjamorim


I think it probably would be valid.

But, instead choosing worst-case samples, I would rather choose samples of different music styles. (If these worst-case samples are "representative" - perfect!)

My rationalization:
If you use samples from different music stiles to tune your encoder, you'll be tuning it for the general public that will be using your encoder.

If you use only problem case samples, you'll be tuning your encoder for listening tests that use only these samples. You'll be tuning it for a limited part of the public that enjoys that kind of music too, but a much wider public would be reached using "generic samples"


Well the idea would of course be that these difficult samples would come from many different genres. smile.gif In fact, most of the time they do. Certainly the ones used to tune the --alt-preset's did.

I guess to further clarify what I'm getting at... if a person says or believes that each of these samples on their own are not representative or good for a valid comparison, what's to prevent someone from saying the exact same thing even if they are grouped up.

"They are all extreme cases.. so who cares?", etc.

QUOTE
[b]After all, what's worth if an encoder performs superbly on Castanets and Fatboy, but fails on Pink Floyd? Such an encoder would be worthless! biggrin.gif


lol! biggrin.gif
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HotshotGG
post Aug 6 2002, 06:41
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QUOTE
After all, what's worth if an encoder performs superbly on Castanets and Fatboy, but fails on Pink Floyd? Such an encoder would be worthless!


All in all if you ask me it's just "another brick in the wall." biggrin.gif


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ff123
post Aug 6 2002, 07:02
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Funny that this topic of sample selection should come up now because I've been beating my head against the wall trying to think of a way to practically, but fairly choose samples, which of course will be much harder in a high bitrate test.

In the extreme (but fairest case), one would just choose randomly from a variety of genres without regard to how "difficult" they are to any of the encoders involved in a comparison. At high bitrates, though, one would probably have to choose many samples to find those difficult enough to discriminate between codecs. Ideally one would like to have (for the sake of argument) a dozen of these difficult samples to say something substantive about relative codec quality. But if only 10 percent (again for the sake of argument) of the randomly selected samples are difficult, there is a little problem -- one would have to go through 120 samples to find the difficult dozen.

120 samples to rate is clearly ridiculous. How does one fairly come up with difficult samples to use in a high-bitrate listening test?

ff123
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Jon Ingram
post Aug 6 2002, 11:10
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QUOTE
120 samples to rate is clearly ridiculous.  How does one fairly come up with difficult samples to use in a high-bitrate listening test?

I imagine that there is a fairly strong positive correlation between a 'challenging sample' for a particular codec, and the bitrate of the encoded file produced by that codec - assuming that you are using a quality based encode (--alt-preset standard, Vorbis -q 6, to name the two which I'm most interested in).

So you should be able to let the codecs themselves do some prescreening for you - if every/almost every quality based encoder uses less than the 'nominal' bitrate to encode that file, then just assume that the file is 'easy', and will not be useful as a discriminator in a listening test.

You are then left with a (hopefully smaller) collection of files which at least some of the encoders consider to be challenging. If not all of the encoders agree, then that's even better (I imagine there will be disagreements between the transform and subband encoders on some samples, which is just what you want).

Of course, you're relying on the encoders being relatively well tuned, and it could be argued that this will bias the sample selection in some way. But you need to be biased in a very similar way in any case, otherwise (say by using a purely random selection) you'll be left with a selection of samples of which 90% will all be rated 5/5.
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Mac
post Aug 6 2002, 11:16
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My own opinion would be that these extreme cases are valid. I listen to the whole of Kalifornia, so being able to encode the first 5 seconds (the fatboy sample) properly would be important to me!

But, I agree that tuning your codec so that it ace's these rare cases at the expense of being the best at the other 95% of samples would be the wrong way to go.

I'm not really sure what factors make a sample difficult to encode. Fatboy seems pretty clear to me, a sound is made of short pulses which our brain percieves as a continuous tone, so that would be a bastard to encode!! What about other things though? If you knew what a general coder found easy and hard, you could just listen to your own cd collection and pick out a great number samples you would imagine to be hard. If enough people did this, you would end up with the hardest parts of every genre and style. Then, you would practically have generic music, but you have cut out all the easily encoded bits, and just have the more taxing stuff, which could then be used to better distinguish between encoders.

My personal expectation of a good coder is to be able to handle snares and symbols properly. Almost all the styles of music I listen to include them, and mp3 often made them sound 'orrible. I might post some samples of the type of thing I mean, but they'd all be aac's or mp3's as I lost the original wav's.. not sure they'd be of any use for testing except to get my point across smile.gif

And to ask my original question again, Frank, where did you get those encoders from? Where would I get them from? smile.gif

[ps. rjam, I disable smileys when I remember to, they still get messed up in quotes tho!]
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Mac
post Aug 6 2002, 11:45
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I wouldn't just favor the samples given a high bitrate by most coders for one simple fact. I like drum'n'bass.

Please, refrain from too much laughter and ridicule :red:


This is come to think of it, a fair complaint against encoders.

Most codecs I've used (lame, aacenc, mppenc) supply a lot less than the average amount of bits to this general type of music. Take for instance:

Roni Size - Brown Paper Bag.
It contains accoustic guitar and high snares (high frequencies are a favourite of roni size), which I'm imagining are among the harder thing to encode and get sounding as nice as the original?

AACenc 2.15 -extreme gives it 158.8kbs
MPPEnc 1.1 --xtreme gives it 144.9kbs
(although LAME gives it justice and gives 194kbs at -extreme)

I would imagine these are all less than average? This has led to files sounding really poor quality because of being given such a low bitrate on lower settings (~normal) I can provide some nasty examples when I have more time. So as I just batch encoded at a certain quality level and then deleted the originals, I later found out shit, I can't trust it to encode at a set quality level. Or at least, my opinion of euqal quality differs from the coders opinion. (and yes, i was stupid to delete the originals before giving the mp3's a proper listening to!)
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Dibrom
post Aug 6 2002, 12:19
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QUOTE
Originally posted by Mac

(although LAME gives it justice and gives 194kbs at -extreme)


LAME really isn't a good example to use for bitrate comparisons. This is simply due to the fact that it will bloat on anything with a lot of high frequency content.. it's not very efficient at all in encoding this.. so it's not really "doing it justice", it's just simply wasteful due to trying to compensate for a design flaw in the format.
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Gecko
post Aug 6 2002, 12:22
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Why should it hurt to tune a codec with extreme cases? I imagine "real music" to be a combination of many of these extreme cases (transient signals, pure tones...) where some are more present than others. So if you have tuned your encoder to handle all the extreme cases well (note that you are tuning all known extreme cases at the same time, without neglecting one in favor of another; I imagine this hard to do), shouldn't it work well on those samples where the extremes are less present (ie normal music, whatever that is)?

I don't know if this holds true, perhaps there is no definite answer. I also think (but can't proove, and probably noone will be able to proove the opposite either) that alot of today's music is very similar sonically. Britney Spears contains many sudden strange noises and synthesizer tones. Destiny's Child use "wicked" percussion (from a drum computer/sampler). And I also believe alot of sound is added in the production process to music that may have once actually been recorded (most of today's rock music).

Heh, so this is today's mainstream music churned out by the producers at 1 hit per minute. Normal music? You decide. I can't really comment on "older" music (I can hear Roberto shouting: Pink Floyd!) since I am not exposed to it as much, but suppose the situation is similar.

Me? I'm a trance head and the places where codecs trip up are just the same as in Britney Spears.

PS: when you say that Frank made an unfair test when he compared castanets encoded with different encoder types and that this sample makes mpc shine and all others suck, well, look at the results again. (Also keep in mind he was using 128k ish bitrates and nowhere did he say he was doing any representative testing)
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Ivan Dimkovic
post Aug 6 2002, 13:56
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QUOTE
Originally posted by Mac

Most codecs I've used (lame, aacenc, mppenc)  supply a lot less than the average amount of bits to this general type of music.  Take for instance:

Roni Size - Brown Paper Bag. 
It contains accoustic guitar and high snares (high frequencies are a favourite of roni size), which I'm imagining are among the harder thing to encode and get sounding as nice as the original?

AACenc 2.15 -extreme gives it 158.8kbs
MPPEnc  1.1 --xtreme gives it 144.9kbs
(although LAME gives it justice and gives 194kbs at -extreme)


Damn, that particular sample is one of my favorite tuning samples smile.gif Now, I can't hear any problems at -extreme (AAC) ?

Also, MP3 needs more bits because of the last scalefactor and limited M/S coding abilities - so it is not good to compare direct bit rate of MP3 and, say, MPC.
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Mac
post Aug 6 2002, 18:33
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Well, I think my hearing is going, because on that song, I can only tell a slight difference between -extreme (158k) and -thumb (65k) ???. But no, I can't hear anything wrong with it at -extreme.

I just wonder why any drum'n'bass tends to be given substantially fewer bits than other music?




QUOTE
I imagine "real music" to be a combination of many of these extreme cases


Hmm, not really. If most music was made up of these hard to code examples, then surely most music would sound pretty awful when encoded at less than 200kbs, like a lot of these samples tend to do!! I only know of two songs that contain anything like the fatboy sample... Kalifornia where the sample comes from, and 18, a song by me (which incidentally, AAC encodes the funny sample bit 30kb higher than the rest of the song)
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Frank Klemm
post Aug 6 2002, 21:23
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QUOTE
Originally posted by Gecko
Why should it hurt to tune a codec with extreme cases? I imagine "real music" to be a combination of many of these extreme cases (transient signals, pure tones...) where some are more present than others. So if you have tuned your encoder to handle all the extreme cases well (note that you are tuning all known extreme cases at the same time, without neglecting one in favor of another; I imagine this hard to do), shouldn't it work well on those samples where the extremes are less present (ie normal music, whatever that is)?

I don't know if this holds true, perhaps there is no definite answer. I also think (but can't proove, and probably noone will be able to proove the opposite either) that alot of today's music is very similar sonically. Britney Spears contains many sudden strange noises and synthesizer tones. Destiny's Child use "wicked" percussion (from a drum computer/sampler). And I also believe alot of sound is added in the production process to music that may have once actually been recorded (most of today's rock music).

Heh, so this is today's mainstream music churned out by the producers at 1 hit per minute. Normal music? You decide. I can't really comment on "older" music (I can hear Roberto shouting: Pink Floyd!) since I am not exposed to it as much, but suppose the situation is similar.

Me? I'm a trance head and the places where codecs trip up are just the same as in Britney Spears.

PS: when you say that Frank made an unfair test when he compared castanets encoded with different encoder types and that this sample makes mpc shine and all others suck, well, look at the results again. (Also keep in mind he was using 128k ish bitrates and nowhere did he say he was doing any representative testing)


Castanets and Fatboy aren't very transient. There's much harder to encode (real world music)
out there where these two are dull and adagio-like. Attack times of some hundred microseconds,
50% of the energy above 10 kHz.

It seems to be that the human ear has the best time resolution above 10...12 kHz.

Currently MPC uses 350 and above kbps for such signals.


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guruboolez
post Aug 6 2002, 21:53
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Frank, I have a question who haunt me for month... I encode my classical music in mpc for month, and I noticed immediatly after leaving mp3 than Musepack had I strange behaviour with some instruments. Piano don't need too much bitrate, with mp3, mpc or Vorbis. But a violon (not a critical instrument ) seems overrated by mppenc : +20% (200 on --standard ; 230-240 on extreme, etc...). Harpsichord, organ... the same thing (a bit less fororgan, but harpsichord is more problematic). With --alt-preset standard, I obtain 180 kb/s, and never reached 200 kb/s : mp3 is very cool for classical listener who don't like Metallica. But with Musepack, Mozart need as much bitrate as AC/DC with mp3 encoding :mad: tongue.gif

I recently find a strange and forgotten instrument, called glass harmonica : an horrible and distorded sound !!! Brrr... With --alt-preset standard, an adagio (quiet but awfull music) need only 150 kb/s ! With mpc --standard : 250 kb/s !!!!

???


Why distorded music (harpischord, baroque instruments) are needing so much bitrate, and why heavy metal don't ? Can you, or someone else, help me to understand this big differences ? Thanks a lot

[sorry for my poor expression, and thanks again for your job]
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